1. Field of the Invention
The present invention generally relates to communication systems, and in particular to a communication system and method for speech information.
2. Description of the Related Art
A widely used one of telecommunications systems is an analog system which uses analog signals to represent information, for example, the classic, voice-based telephone system which is being replaced by the digital communications system. Since digital signals are easily stored and copied, various services including voice message can be provided.
In the case of digital voice communications, transmission of a large amount of voice data may cause delay during interactive conversation between users. In general, users do not tolerate appreciable delay. To overcome such a delay problem, there have been proposed several systems employing speech recognition and speech synthesis techniques.
In Japanese Patent Unexamined Publication No. 60-136450, the digital speech data is converted into character code information by means of speech recognition technique. The character code information is transferred to the destination where the received character code Information is converted back to speech data by means of speech synthesis technique. Since the amount of character code information is munch smaller than that of digital speech data, the real-time interactive conversation may be achieved.
However, since the character code information is converted to the speech data by the speech synthesizer at the receiving side, the same voice reproduced even if different users speak at the sending side. Therefore, the receiver cannot know who is calling. Further, it is difficult to know that a calling party is a different person using a registered name or password.
An object of the present invention is to provide a speech information communication system and method which can easily identify who is calling.
Another object of the present Invention is to provide a speech information communication system and method which can screen calling parties to desired registration data.
According to the present invention, an input speech signal is converted to character code information and speech feature information which are transmitted to a receiving side where they are combined to produce an output signal. In other words, a first converter converts an input speech signal to character code information and speech feature information. The character code Information and the speech feature information are transmitted to a transmission line. A receiver receives character code information and speech feature information from the transmission line and a second converter converts received character code information to an output signal depending on received speech feature information.
According to another aspect of the present invention, the second converter may convert received character code information to an output display signal for displaying the received character code information on screen depending on received speech feature information.
Since the character code information and the speech feature information are both transmitted to the receiving side where they are combined to produce the output signal, the amount of transmission data is reduced and a receiving user can easily identify who is calling.